Tips & Tricks

Technique : Effects / Processing

The way you use effects and processors can make or break a mix. Paul White offers 20 useful tips to help you get it right first time.
Recording can be fun, but for me, the most rewarding part of any
project is doing the final mix. It's at this stage of the proceedings
that effects and signal processors can be used to turn a simple
recording into a major production -- but it's also easy to overdo things
and spoil the end results. This month I've put together 20
easy-to-remember tips that will allow you to control your effects units
rather than vice versa. And so, without further ado and in no particular order of importance:

1. Reverb creates the illusion of space, but in doing so it also
'smears' the stereo localisation of the original sound source, just as
it does in real life. If you want to maintain a specific stereo
placement for one or more sounds in a mix, consider using a mono reverb
effect and panning the reverb to the same position as the original dry
sound.

2. Reverb is very useful for making vocals sound more musical
and for making them sit with the rest of the mix, but adding too much
will have the effect of pushing the vocals back, rather than allowing
them to take front position. Experiment with pre-delay values of
60-100mS to help counter this, and also try using a reverb patch that
has a lot of early reflections, as these help reinforce the dry sound.
You can learn a lot from listening carefully to records you like to see
how much and what type of reverb is used. Often it's rather less than
you think.

3. Bright reverbs can flatter vocals, but may exaggerate
sibilance. As an alternative to de-essing the vocals, try instead
de-essing the feed to the reverb unit, so that sibilance is removed
before the reverb is applied.

4. Reverb is probably the most important effect in the studio,
so don't compromise by using a low-quality software reverb plug-in just
because you're short of processing power. Use a good external hardware
reverb unit if you have one, otherwise choose a more powerful software
plug-in to treat the vocal track in non-real time. This may involve
off-line processing or doing a real-time 'bounce to disk' of the vocal
track in isolation, via the plug-in.

5. Vocals almost always require compression, but rather than
doing all the compressing at the recording stage, apply a little less
compression than you think you might ultimately need, then add further
compression when you come to mix. This dual-stage process ensures you
don't record an overcompressed sound, whilst still allowing you to even
out the level of the recorded signal.

6. Compressors bring up low-level noise just as effectively as
they do low-level signals, so try to gate the signal prior to
compression when you're mixing. Also, use no more compression than you
need, or the signal-to-noise ratio may be compromised unnecessarily.
However, it's usually unwise to gate the compressor input during
recording for the reasons explained in the next tip.

7. Avoid gating during recording if at all possible, as a badly
set gate can completely ruin an otherwise good take by chopping out
low-level sections of the wanted audio. Instead, gate during mixing,
when you have the chance to reset the parameters and try again if it
doesn't work out first time. A further benefit of this approach is that
any noise, crosstalk or spill accumulated during recording will also be
gated out.

8. Always gate signals prior to adding reverb if you can --
gates can easily chop off the tail end of a long reverb. Furthermore, if
you add reverb or echo after gating, any minor gating artifacts may be
completely

"It's at the mix stage
of proceedings that effects and signal processors can be used to turn a
simple recording into a major production -- but it's also easy to overdo
things and spoil the results."

hidden by the natural decay of the reverb or echo.
Any noise added to the mix by the reverb unit should be negligible
providing you've paid attention to gain structure and level setting when
adjusting the effects.9. Don't always set your gate to fully attenuate the signal when
the gate is closed. In some situations, it may sound more natural if a
low level of background sound is still audible between wanted sounds,
and when working with drums, you'll find the gate opens faster if the
range control is set to around 12dB rather than to maximum.

10. Single-ended noise-reduction units (the type that work by
applying level-dependent top-cut) can be very useful in reducing the
perceived level of hiss during material where there are no silences that
would allow a gate or expander to operate. However, make constant A/B
comparisons to ensure that there's no obvious top-end loss when the unit
is switched in. If there is, lower the threshold slightly until you get
an acceptable compromise between high-end loss during low-level
passages, and audible hiss. As with gates, applying reverb after dynamic
filtering may help disguise any side-effects as well as safeguarding
the reverb tails from being truncated.

11. Don't add long reverb to bass sounds unless you have an
artistic reason to do so, as this tends to muddy the low end of the mix.
If you need to add space to a kick drum, try a short ambience program
or a gated reverb as an alternative. If you are in a position where you
need to apply reverb to an entire drum mix, roll off the low end feeding
the reverb for a cleaner sound.

12. Chorus is a useful effect for creating the illusion of space
and movement, but it also tends to push sounds back in the mix, rather
as reverb does. If you need a sound treated with chorus to stand out in a
mix, try either panning a dry version of the sound to one side and a
chorused version to the other, or ensure that the song's arrangement
leaves plenty of room for the chorused sound.

14. Equalisation is often used as an alternative to getting a
sound right at source, but the result is seldom as satisfactory as doing
things properly. Nevertheless, on occasions where equalisation is
necessary, applying cut to the over-emphasised frequencies rather than
boost to weaker ones generally results in a more natural sound,
especially with vocals and acoustic instruments. This is especially true
of in-desk equalisers or budget parametrics, as they often sound nasal
or phasey when used to boost mid-range sounds.

15. Sounds can often be made to sit better in a mix by
'bracketing' them with high- and low-pass filters so as to restrict
their spectral content. Many console EQs don't have the sharp filters
necessary to do this, but the side-chain filters fitted to many gates
are often ideal for the job. Simply set the gate to its side-chain
listen mode, then use the filters to shave away unwanted high and low
frequencies. Acoustic guitars often work better in a mix if the low end
is rolled off in this way, though the high end can usually be left
alone.

16. When setting up a mix, try to get the mix sounding close to
right before you add any effects or signal processing. Once you've got
this right, add further vocal compression if needed and also apply just
enough reverb to make the vocals sit comfortably with the backing track.
When you're happy with the overall timbre and balance, adding effects
for 'effect' should be easier. Remember that, in most cases, effects are
there just to add the final gloss -- they won't compensate for a poor
balance or bad basic sounds.

17. Still on the subject of effects in the mix, don't be tempted
to hide poor playing by heaping on more effects, it never works -- take
it from someone who's tried everything at one time or another! However,
thanks to the wonders of modern technology, slightly imperfect vocal
pitching can be tightened up almost magically using pitch-correction
processors, such as Antares' Autotune software or ATR1 hardware.

18. Go easy when using enhancers to treat complex signals such
as a whole mix as it's very tempting to go too far. Make frequent use of
the bypass button to remind yourself just how radically the sound has
changed, and if you're adding more than a little high-end enhancement,
check the bottom end to see if that needs bringing up to keep the
overall mix in balance.19.
Often it's better to enhance just some elements of a mix so as to make
them stand out from the rest. The best way to do this is to connect the
enhancer to a pair of group insert points, then send all the sounds that
need enhancing to that group. Listen carefully to enhanced vocals as
the process can often exaggerate sibilance problems.

20. Treatments designed to increase the stereo width of a mix
(other than the simple
mixing-antiphase-signals-into-the-opposite-channel trick) can have
detrimental effects on mono compatibility. Use your console's mono
button to check that your mix doesn't lose too much when it's played in
mono, as this is important when material is played over mono radios or
TVs. Listen to see if the subjective balance or timbre changes by an
unacceptable degree. If it does, either use less overall width expansion
or leave the main mix elements untreated and only process secondary
sounds, such as incidental percussion, sound effects, effects returns
and so on.

Friday, December 27, 2013

Tips & Tricks

Technique : Effects / Processing

Antares Auto-Tune is a powerful pitch-correction tool which is already an industry standard for tightening up vocal performances. As Paul White explains, however, it has the potential to do much more...

Antares Auto-Tune is, in my
view, one of the few truly innovative musical developments of the past
couple of years. Other devices may employ more novel technology, but Auto-Tune
provides a practical answer to the very real problem of pitching
imperfections, via an easy-to-use box or software package. We've
reviewed both hardware and software versions of Auto-Tune in SOS already, though there's a more recent addition to the series in the shape of Auto-Tune LC,
a low-cost VST plug-in that offers all the functionality of the
hardware version (with the exception of external MIDI control) for under
£100. A DirectX version of Auto-Tune is also available for those
working on the PC platform, so just about anybody who needs pitch
correction can now access it affordably. What's more, at least two other
companies are planning hardware or software products that do the same
kind of job (indeed, TC Electronic's Intonator is reviewed in this issue
of SOS, starting on page 198), so I foresee this being a growth area.

The main function of Auto-Tune
is to tighten up vocal pitching, something it does very well, but I've
since experimented with the system and found that it can also be used to
produce creative effects and to improve monophonic instrument
performances. This article is not a re-review of Auto-Tune, but rather a look at some of its less obvious applications.

Auto-Tune Theory

Before setting out to use Auto-Tune, it's useful to have a general idea of how it works. In order to correct pitch, a system such as Auto-Tune
needs to be able to read the pitch of the incoming signal. This is only
really possible with monophonic sources at present - the algorithm was
apparently a spin-off from work in analysing seismological data, and it
seems well adapted to tracking the pitch of the human voice.

Once the pitch has been detected, Auto-Tune
then uses pitch-shifting techniques to change the pitch of the original
sound to match the nearest note in a user-definable scale. It is
possible to leave Auto-Tune set to a chromatic scale (ie. one
containing all notes), but then it's likely the pitch of a badly sung
note could be corrected to an 'illegal' (out-of-key) note because that's
what it happened to be nearest to. Far better to input only those notes
the piece of music is using - then the worst that can happen is that a
horrendously badly pitched note will be corrected to the wrong note in
the right key!

Forcing incoming notes to the right
pitch is only part of the story, because if that was done too
efficiently, you'd end up with a very flat, pitch-quantised vocal
performance. Fortunately, there's a slider to adjust the rate at which
correction takes place: by adjusting this carefully, natural bends and
vibrato are allowed through unaltered, but as soon as the input settles
on a specific note for any length of time, it's pulled into perfect
pitch. This was demonstrated most impressively at the Frankfurt
MusikMesse by demonstrator Gerry Basserman, who used a theremin as the
input source. With the tracking set to fastest, the theremin sound was
changed so that it would only produce discrete, stepped pitches, just
like a keyboard. However, setting a longer correction rate allowed Gerry
to play the instrument apparently normally - except that when he
settled on a note, it always slid into perfect pitch. As he himself
confesses, this made him sound a much better theremin player than he is -
though to be fair, he's actually pretty good anyway!

Creative Uses

Because Auto-Tune doesn't use
any kind of formant correction, any sound pulled far from its original
pitch starts to sound unnatural in the same way any conventionally
pitch-shifted signal does. In normal use, this isn't a problem, because
the usual amount of pitch correction is less than half a semitone, but
the effect can be abused in creative ways by deliberately setting large
intervals. For example, you could set up a target scale containing only
octaves and fifths, then sing a load of made-up 'pseudo-ethnic' nonsense
to see what comes out the other end.

In this instance, the voice timbre is
transformed proportional to the pitch difference between what you're
singing and what the nearest scale note is set to. For example, you can
sing a steadily rising tone and there will be no pitch change until you
got within range of the next note in the target scale. However, the
timbre of the fixed note will change as the pitch shifter works harder
and harder to correct it. In this example, the pitch-shifter will
attempt to push the pitch of the note down to compensate for you singing
higher, so the timbre will take on a darker, bigger quality. Then, when
you get within range of the next note in the scale, the timbre will
flip as the pitch-shifter tries to push the pitch upwards. Gerry worked a
few examples of this type into his demo and I have to say the resulting
'pseudo-world music' was most convincing.

Experiments

I was recently asked to do an album of music themed around the solar eclipse, and this provided the perfect excuse to use Auto-Tune
for something other than vocal correction. For one of the tracks, I
wanted to use a wooden North American Indian flute I'd picked up on my
travels, but there were two problems - I couldn't really play the flute
very well, and the notes it produced were just far enough off concert
pitch to be irritating. The first problem I got around by a mixture of
practice, bluffing and hard-disk editing, while the second required the
software version of Auto-Tune programmed to the same pentatonic
scale as the flute. I set a slowish tracking time so all the note bends
and trills would get through unaffected and let nature take its course.
The result was a perfectly natural-sounding flute part with all the
sustained notes nicely in tune - it even surprised me! This was the
first time I'd tried Auto-Tune on a source other than a voice, but the tracking algorithms seemed to have no difficulty at all dealing with it.

The next test was to try Auto-Tune
on a slowish lead guitar part that featured a lot of 'long' string
bends and tremolo arm tricks. Though I don't play guitar nearly as much
as I should, my pitching while bending is normally pretty good, but a
slowish pitch-correction setting on Auto-Tune really added polish and precision to the performance. On the occasions that I picked double notes, Auto-Tune
just ignored me - it only seemed to take effect when I was sustaining a
single note, and of course the slow pitch-correction setting allows any
vibrato to come through normally. Considering the complexity of a
distorted guitar waveform, Auto-Tune tracked it very well with no glitches or warbles. I also conscripted Paul Farrer's help to test Auto-Tune on some cello parts he was recording, and he reported the experiment a great success.

"The more you think about
it, the more applications there are for this amazing tool - using it
just to polish up vocal tracks really is under-exploiting its true
potential."

When Cher featured that distinctive vocoder effect on her 'Believe' single, a lot of us thought that it had been done on Auto-Tune, though as the February '99 SOS article revealed, that proved not to be the case. Nevertheless, you can get a very good approximation of that same effect from Auto-Tune
simply by setting the pitch-correction rate to its fastest setting and
entering the appropriate scale for the song. I tried it as a bit of a
joke and found it uncannily like the real thing - it's a shame that
after just one outing, the effect is already a cliché to avoid. A
side-effect of using very fast pitch correction is that shallow vibrato
is ironed out altogether, while deeper vibrato turns into a trill.
Normally you wouldn't want this to happen, but when you're being
creative, little accidents like these can produce very usable results. Auto-Tune
also has the ability to add delayed vibrato to whatever sound is being
processed, so in theory, you can strip out the vibrato from the original
performance and replace it with something far more mechanical and
precise.

Things to try

Auto-Tune is so effective that
I'm always adding to the mental list of things to try out with it.
Monophonic slide guitar should be easy enough to fine-tune, but then
there's the option of speeding up the pitch correction and turning it
back into a chromatic instrument. Set a musical scale and you have the
makings of an electric slide dulcimer!By the same token, those who can play
just a little fretless bass, cello or violin might find that their
efforts become a lot more musically useful after a trip through Auto-Tune.
Certainly Paul Farrer's experiment showed that a reasonably well-played
cello part could be made to sound much more precisely pitched without
the result appearing to be processed.

One function of the ATR1 (the hardware version of Auto-Tune)
I haven't mentioned yet is its ability to shift pitch according to a
MIDI input. In theory, this means you can input a MIDI melody and
whatever audio input you have fill be forced to fit that melody. With
careful setting up, you can get very natural results, but how about
striving for the unnatural by making the human voice sing impossible
arpeggios or taking a single-note instrument like the digeridoo and
forcing it to play a melodic bass line? The more you think about it, the
more applications there are for this amazing tool - using it just to
polish up vocal tracks really is under-exploiting its true potential.

Tips & Tricks

Technique : Effects / ProcessingIn the final part of his short series on pushing back the boundaries of effects processing, Paul White
explores many different applications of audio filters, as well as
exploring the possibilities of granular synthesis. This is the last
article in a two-part series. Read Part 1.

One effect that has migrated from the
synthesizer world is filtering. The simplest synth filter has a low-pass
response with variable resonance at the cutoff point, however, there
are a number of other filter types which can be useful, and details of
all the different filters can be found in parts four and five our Synth
Secrets series, in SOS August and September 1999 respectively.
Just running audio through a synth-type filter can be fun, particularly
when you experiment with higher resonance settings, but things get more
interesting when you manipulate a filter in real time. An envelope is
often used to do this, and the trigger that this envelope requires can
be derived from the audio signal itself.

The least complicated approach is to
use the envelope generator to control the filter's cutoff frequency, and
to cause the envelope to trigger whenever the incoming audio signal
exceeds a certain threshold. This works well when sounds to be processed
are separate and have clearly defined starting points, but is less
effective for sounds that overlap, as triggers can become less reliable.

Remember that the envelope above (or a
different one triggered in the same way) can also be routed to control a
filter's resonance, for extra variety. What's more, other modulation
sources can be used in place of an envelope generator. LFOs can be fun
in this role, either used free-running or with their rates sync'ed to
tempo, and envelope followers can also be used. The latter could control
the filter cutoff frequency according to the level of the input signal —
in other words, the higher the input signal level, the higher the
filter frequency (or vice versa if that's how you want to set it). The
sound this creates can often be musically useful, though where the
initial sound is modulated in level, the resulting undulating envelope
can lead to the filter opening and closing in a unmusical and seemingly
erratic way.

Where the incoming sounds
are from a sampler or a MIDI synth, then MIDI triggering and control of
any audio-processing filter can be employed, and clearly this will work
consistently whatever sound is being processed. However, it is
important to note that, in a typical synthesizer, each voice will have
its own filter while, in the case of an external filter module, all the
voices will be processed via the same filter — this means some decision
has to be made as to how the single processing filter will behave when a
new MIDI trigger is received. Should the filter envelope start again as
each new note is played or should it only reset after all keys are
released? Some units make this choice for you while others provide
switchable functions. Other points to note are that stereo filters are
necessary for stereo signals and that filters should be inserted into
the signal path, rather than being used in a send/return effects loop
configuration.

Morphing

Although there are a
few genuinely innovative effects available, most effect treatments are
based on combinations of distortion, filtering, delay, reverb and
pitch-shifting. One way to get new mileage out of these old effects is
to make them dynamic — use MIDI controllers to change the character of
the effect as the sound progresses. For example, what starts out as a
flange might end up as a simple repeating delay — all you need to do is
reduce the modulation depth, increase the delay time and reduce the
amount of feedback over the space of a few seconds. Most sequencers with
graphic editors make this kind of thing fairly easy, but there are also
dedicated hardware processors capable of creating morphing effects. Some
have morphing built in as a specific function, while others allow you
to assign a number of parameters to a pedal, then change them all
simultaneously during performance — as the pedal position is moved, the
parameters change and the effect morphs. Sometimes you'll find that what
happens in the middle of the transition is more interesting than what
happens at either end, so allow yourself time to experiment if this kind
of thing appeals to you.

Stand-alone filters can
be used in a number of different contexts to add interest and movement
to a sound. An obvious application is to take an otherwise filterless
synthesizer (such as a Kawai K1, Korg Wavestation or Alesis Quadrasynth)
and treat its output with a filter. It's also very common to use
filters to process sample loops and, in this application, the automation
offered by many software plug-ins and MIDI-controllable hardware units
is very attractive. Complex, tempo-related effects can be created within
your sequencer and then copied to as many bars as are required.

More Fun With Filters

We've had resonant high-pass, low-pass
and band-pass filters ever since the first analogue synths, and though
they're still widely used today, there's a lot more that can be done
using more sophisticated filter types. We've already seen how you can
use the complex filtering of a vocoder to create interesting new sounds,
but that is only the start. There is a great deal of mileage to be had
from processing different frequency bands separately (see 'Divide &
Conquer' box on page 124)

One particularly complex multi-band filtering effect is available within Emagic's Logic Audio — the Spectral Gate
plug-in. Although the documentation doesn't make it clear exactly how
the filters are configured, a few minutes playing with the controls
demonstrates the range of effects that can be created: everything from
the more obvious filtering characteristics to sounds that appear almost
resynthesized. It's very easy to create metallic, electronic-sounding
textures from quite conventional input material, though, as with all
filtering, the more harmonically rich the input, the more interesting
the output.

While on the subject, it's also worth trying out the 'Convolve' process in Bias Peak
or any other software that offers the facility. This sounds to me like a
filter-based effect, and it allows the characteristics of two sounds to
be merged in order to produce a new sound sharing characteristics of
both. This process is often used in sound design to combine sounds in
unusual ways.

breveR & yaleD!

Conventional delay and echo effects
are a mainstay of music production, though last month I suggested a few
ways to make these more interesting. A further way of adding interest to
effects is afforded to us by the recording process, which enables us to
make use of negative time. By this, I mean effect sounds which are
audible before the sound that they process even starts playing —
something that the physical laws of real life don't allow, unless in
close proximity to a concentrated source of tachyons! Once something is
recorded, temporal rules can be broken. Sounds can be reversed, they can
be treated with reverb or delay that starts before the sound itself, or
reverse reverb can be added to a 'right-way-round' sound. This latter
trick used to be popular during the '70s for music production and is
still used extensively in film work to create demonic voices.

When analogue tape was
the standard recording medium, reverse reverb was accomplished by
playing a tape backwards, feeding the desired track through a reverb
unit, then recording the reverb to a spare track. Once the tape was
replaced on the machine the correct way around, the reverb track would
start playing before the track it was derived from, with the reverb
sound's envelope building up slowly in a suitably eerie manner. The same
trick can be achieved in a tapeless environment (such as within a MIDI +
Audio sequencer) by reversing a section of audio, adding reverb or
delay, bouncing the processed result to a new track, then reversing both
tracks again. I covered this procedure in some depth in SOS
December 1998, for those who'd like to try it. For a more ambitious
project, try one of the chopping techniques described last month, using
reversed sections of sound or effect for some of the segments.

Of course, the other thing that's very
easy to do in a tapeless environment is set up a conventional reverb,
record this to a spare track, then slide it a beat or two ahead of the
track being processed. This produces reverb that's the right way around,
but which still comes before the sound that supposedly created it — you
could almost think of it as negative pre-delay!

Divide And Conquer

A further avenue worthy of exploration is
multi-band processing. If a crossover type of filter is used to separate
the audio being processed into three frequency bands, roughly
corresponding to bass, middle and treble, then each band can be
processed separately. Applying distortion to the individual bands is
quite instructive as the sound holds together much better than when
distorting the entire signal all in one go. At lower levels of
distortion, the signal sounds compressed and more energetic, making this
a good choice for treating drum loops, and even when the sound starts
to get noticeably crunchy, it still retains enough integrity to be
useful. Try this process on bass sounds as well as drums, and also try
varying the amount of distortion in each frequency band. The Steinberg Quadrafuzz
VST plug-in offers four-band distortion and demonstrates this principle
very nicely. Another multi-band processor to check out is the Electrix
MoFX, which provides a distortion block, and a three-way crossover to
which flange, tremolo and delay effects can be applied in various ways.If you're using a system such as TC's Spark
that allows you to combine VST plug-ins rather more flexibly than most
systems, you could try applying different processes to the various
frequency bands. For example, compress the high end to get more
high-frequency energy, distort the bass end for more deep-down power
and, while you're at it, add delay or modulation to the mid-range. It is
also worth placing a high-cut filter after the bass-band distortion to
prevent too many harmonics in the mid-range. The block diagram below
illustrates this configuration.

Granulated & Lumps

One of the current buzz words in sound
creation is 'granular' synthesis, which simply means taking very small
slices of sound, often from very different sources (or from different
times within the same source), and then joining them together to create a
new sound. The difference between granular synthesis and wavetable
synthesis seems to be mainly the length of the individual segments used.
Granular segments may only be a few tens of milliseconds in length.
Most of the results I've heard from granular synthesis remind me of a
day out at the dentist, but for those on the cutting edge of techno, it
might be exactly what is needed. I don't know of any granular effects
boxes, but it should be possible to fake your own using a sequencer if
you have the patience and a suitably sadistic mind. There are also
software packages that can help, such as CDP's GrainMill.

At its simplest, you could set up two
audio tracks with the level automation programmed to gate the sound on
and off very rapidly. This is easiest to achieve using your graphical
editor to draw nice neat square waves linked to MIDI continuous
controller seven. By arranging it so that one channel is off while the
other is on, and vice versa, you should end up with something very
crunchy indeed. You should probably be aiming for something like 64
transitions per bar or more, so make use of the copy and paste
facilities of your sequencer when creating the granular 'chopping'
templates. When you get a good one, save it in your default song so that
you can use it with other sounds. If you have a plug-in filter that
offers a sample and hold facility, you can get some pretty
granular-sounding effects out of it if you can persuade the
sample-and-hold rate to go up high enough. If you can get it fast enough
so that the individual steps blur into a continuous noise like somebody
cleaning their oven with an angle grinder, you've probably got it about
right!

Another way to approximate a granular
type of effect is to use the manual digital scrubbing facility on a
piece of recording hardware or software, and then record the results.
Often you'll find there are two scrubbing modes, one that works rather
like tape varispeed and another that constantly plays back a tiny loop
of audio as you move through the audio file. It's the second one of
these you need for real granular-style emulation — just move through a
sound slowly and linger on any sections that sound particularly
interesting. Again, sample anything that sounds useful.

Do You Believe In Life After Cher?

Intonation correction such as that provided by Antares Auto-Tune
was never designed as an effect but, by adjusting the parameter
settings, you can still coax some very interesting sounds out of it. The
famous vocal sound on Cher's 'Believe', for example, may be evoked
using Auto-Tune, just by setting it to correct at the fastest
possible speed. This effectively quantises the vocal pitch to the
nearest note in the selected scale, and is becoming a more and more
fashionable sound with each passing Top 40. If you're wanting to try
this yourself, be aware that anything more than mild vocal vibrato can
cause unwanted trilling. Not to say that this can't be turned to your
advantage and used creatively: the trills or slurs (depending on the
correction speed chosen) produced by heavy vibrato and pitch-bending can
be great for adding a hint of eastern promise. What's more, this trick
isn't only useful for vocals — it can also be great for lead guitars and
synth solos.Pitch correctors can
also get really fun when they allow you to feed MIDI note information
into them from a keyboard or sequencer, so as to force sung notes to new
pitches. With a fast correction speed, this creates the familiar
quantizing effect, but because the degree of pitch-shift depends on the
difference between the original note and the target MIDI note, you may
also hear more obvious pitch-shifting artifacts, which can be used
creatively. Using a slower correction speed causes the vocal or
instrumental line to portamento to the new pitch — an effect which can
be good within pseudo-world music.

There are some spectacular granular treatments in Native Instruments' Reactor and Dynamo that
work on short samples of sound, so you can load in your own audio clips
and mangle them mercilessly before recording the results as a separate
audio file. It's more of an instrument than an effects processor, but if
you get the results you want, does it really matter? What's more,
because VST 2 virtual synths can be automated, you can set up your
granular effect within a song and know that it's always going to come
back sounding the same each time.

Coda

While there may not be as many new
effects as we'd like, it's often possible to create some
extreme-sounding treatments by either combining existing effects or by
making existing effects dynamic in some way. Those reclusive people who
design the sounds for sample CDs often set up huge loops of sound
processors and delays that feed back on one another, then they record
the output and select the best bits. If you have a few pieces of
outboard, give this a try using lots of delay feedback so that
everything is just below the point of breaking into self oscillation.
Sometimes you don't even need an input signal to set the whole thing
off!

By creating repetitive changes that
happen too fast for the human ear to perceive, it's easy to emulate
granular synthesis or complex modulation where the result is often
dissonant and mechanical sounding. What counts is that the result is
musically useful, and even treatments that result in atonal mayhem can
sound good in context, especially when used as part of a rhythm track.
Don't rule out cheap effects pedals either, as some of these produce
surprisingly musical sounds, even if they don't have the best technical
spec. My best advice is to break a few rules. If a box says 'guitar
processor', try it out on vocals or drums to see what it does. You have
nothing to lose but your sanity!

Thursday, December 26, 2013

Tips & Tricks

Technique : Effects / ProcessingMost
people are familiar with basic reverb, delay and modulation effects,
but what lies beyond? In the first part of a new series, Paul White explores the twilight zone of effects processing. This is the first article in a two-part series. Read Part 2.

It's often said that there's little to
be had in the way of novel effects nowadays. Most effects are either
standard reverb, delay, modulation or pitch-shift, but that doesn't mean
that there aren't other effects to be found lurking in the dark corners
of your multi-effects box or software plug-ins folder. Some of the more
bizarre effects have been around for years — for example, vocoders,
ring modulators and chordal resonators — but the freedom provided to
designers by the newer plug-in formats means that more off-the-wall
stuff is appearing all the time. The aim of this article is not to
concentrate too much on specific products, but rather to explain some of
these less common effects types and to make a few suggestions about the
ways in which they can be used.

Exterminate!

Ring Modulators are intriguing devices
designed to process two input signals in such a way that the sums and
differences of the input frequency components are generated while the
input signals themselves are suppressed. For example, if you were to put
in two sine tones at 500Hz and 600Hz, the output would comprise tones
at 1100Hz and 100Hz. Conversely, feeding the same 500Hz tone into both
inputs would produce components at 0Hz (a silent DC offset) and at
1000Hz (an octave up from the pitch at the inputs). However, the results
are only as simple as this when you input pure tones — when
harmonically rich sounds are used, all those harmonics contribute to the
sum-and-difference process, resulting in a harmonically very complex
output. Note that an output will only be produced from a ring modulator
when signals are present at both inputs, so if level fluctuations are a
problem then it may be worth compressing one or both input signals.

Because of the way in
which the output frequencies are generated, ring modulators generally
produce atonal, non-musical sounds, which has made them popular for
science-fiction special effects, but they can also be used musically
with a little care. For example, if two similarly pitched synth patches
are ring modulated together then, providing the input waveforms are not
too harmonically complex, the output can be both interesting and
musically useful. Some non-harmonic components will almost certainly
still be present, and detuning the two inputs by a very small amount can
produce unusual low-frequency beating effects, but you can arrive at
some very worthwhile sounds in this way. If you find ring modulation a
little too strident for you, it can often be made more palatable by
blending some of the original unprocessed input in with the processor's
output. If you're still a bit cagey about using this effect, then
perhaps the safest tactic is to use it as a sound design technique,
sampling any isolated moments for later use — ring modulating a 100Hz
tone with a vocal to produce the familiar Dalek voice is always fun, at
least!Processing percussion via a ring
modulator can be good — use a pitched synth sound for the other input
and you'll end up with a metallic, pitched drum part that could form the
basis of an experimental electronic song or dance track. Ring
modulating different cymbal sounds together is also an interesting
experiment, which creates new, electronic-sounding cymbals.If you want to create new sounds and
treatments based upon a basic ring-modulation sound, try combining it
with other effects. For example, use a dry sound as one input to the
ring modulator and its reverb or delay as the other. You can also
further process the ring modulator output using conventional but
dramatic effects such as flanging or heavy delay.

Spark's Magic Piano

At one time, vocoders were considered
quite esoteric, but nowadays they come built into some multi-effects
units — less costly stand-alone units are also fairly common. On top of
that, there are some very effective vocoder plug-ins that can be used
within sequencers.

Like the ring modulator, a
vocoder requires two inputs to generate an output, and to make this
process clear, a block diagram is shown in Figure 1. Essentially, the
vocoder superimposes the frequency spectrum of one sound (called the
modulator) on a second sound (known as the carrier). The way this is
achieved is that the frequency spectrum of the modulator is continually
monitored using a bank of frequency-spaced band-pass filters, and the
information used to control the gains of a corresponding bank of
band-pass filters in the carrier's signal path. Thus, as the spectrum of
the modulator changes, the carrier's filter bank settings follow it. If
a voice is used as the modulator and a harmonically rich musical sound
as the carrier, this results in the classic vocoder sound — the voice
seems to take on the pitch and timbre of the carrier sound, but the
vocal articulation is still recognisable, because of the dynamic action
of the filter bank following the continually changing spectrum of the
voice. As you might imagine, the more filter bands the vocoder has, the
more accurate and intelligible the speech-like element of the output
signal.

It is apparent from this description
of the vocoding process that you need signals arriving at both inputs
simultaneously before you can obtain an output signal. It can also be
helpful to compress both inputs to the vocoder, in order to keep the
output levels stable. On the other hand, if the modulation input is a
voice, you might find that the vocoder is triggered undesirably by
breath noises, in which case a gate inserted between the microphone and
the vocoder's input will also be an improvement.

The talking synth effect has been used
on countless records (for example, 'Blue Monday' by New Order, 'Mr Blue
Sky' by ELO and 'Rocket' by Herbie Hancock), but this isn't the only
way to use a vocoder. By substituting the vocal input with a recording
of background noise in the local pub, and by vocoding this with a rich
synth pad, you can create a very organic pad sound with a lot of
movement. Similarly, two different synth sounds or samples can be
vocoded together to create a totally new sound. If the modulator signal
includes dramatic changes, such as a filter sweep, these will be imposed
on the carrier. The real key is to experiment, but a point to keep in
mind is that, because the end result is created subtractively, the
carrier signal needs to be harmonically rich in order to give the
filters something to work on. If the carrier is a synth sound, an open
filter setting combined with a sawtooth or pulse wave works well.

When vocoders were first developed, it
was realised that, while the pitched elements of vocal sounds provided a
good modulation source, unpitched vocal components such as 'S', 'F' and
'T' sounds tended to get lost, and so vocal clarity was lost. Different
strategies were devised to help with the intelligibility of vocoded
speech. One of these was to replace some of the consonant sounds with
bursts of noise, but a far simpler method was to add a high-pass
filtered version of the vocal input into the output. The latter method
works well, because the higher-frequency region of the vocal spectrum
contains most of the energy of many vocal consonants, yet without many
pitched components. By filtering out everything below 5kHz or so, then
adding the remaining high frequencies to the vocoded signal, vocal
intelligibility can be improved enormously. If this facility isn't
already included in your vocoder, it can be patched up using hardware,
or within a plug-in environment that allows series and parallel routing
(such as in TC's Spark).

Further sophistication is offered by
some advanced vocoders where it is possible to swap around the filter
bands, such that the level of the modulator input at one frequency can
be mapped to control a filter band at a different frequency. Though few
dedicated vocoders have this function and there's no simple way to fake
it, implementations are possible using a software modular synth and it
can really open up the gates of weirdness.

Resonators

Almost everyone has tried setting a
DDL to a very short delay time and then increasing the feedback. The
result is a 'ringing' sound at a pitch determined by the delay time,
particularly when it is excited by percussive sounds. For example, a 1mS
delay will produce a resonant pitch with a fundamental of 1kHz, 10mS
will produce a resonant pitch with a fundamental at 100Hz, and so on.
With drums, it produces an effect not unlike playing them in a resonant
brick tunnel — check your train timetable, though, before trying this at
home!

A similar effect can be
created by setting up a band of equalisation with high boost and
resonance values — this will ring at whatever turnover frequency you
select. If you have a number of bands available then you can set up a
number of resonant peaks, which already holds a lot of scope for
experimentation. However, if you can automate the frequencies of the
resonant peaks, then you can get really creative.

Lexicon used the resonator principle
in the PCM80 to create their resonant chord effect, and something
similar was used by Alesis in their original Quadraverb. The concept was
that MIDI note information from a keyboard or sequencer could be used
to tune one or more resonators to specific musical notes so that any
input signal passed through them would cause the filters to ring or
resonate at musically relevant pitches.

Percussive sounds seem to work best
with resonator programs and, because of the effect of the resonators and
their musical pitches, the drum sound becomes more abstract and
harmonic. Considering how dramatic this effect can be, it's surprising
that it doesn't feature on more records, though I know that a number of
sound designers use it for creating less conventional rhythm loops. If
you have a multi-effects unit with a MIDI controllable resonator, either
monophonic or polyphonic, I'd recommend you try it out at least once so
that you get to know the extent of its capabilities. It can help to
make the effect more obvious if you increase the resonance or feedback,
but otherwise I don't have many tips for experimenting with this effect
other than 'suck it and see'!

Chopping & Changing

I'll finish off for this month by
looking at some of the triggered gating effects that can be set up. It's
fairly well known that a gate can be triggered via its side chain to
chop up audio in a rhythmic way, but it's sometimes possible to take
this concept a little further. A MIDI gate or plug-in is easiest to use
for this purpose, as you can feed in a rhythmic sequence of
note-messages to trigger it. However, any regular gate with an external
key input can be triggered using a fast-attack, fast-release synth tone.

Now let's look at some
ways of making rhythmic chopping more interesting. One option I've heard
used to great effect on vocals is to set up an even tempo-related chop
at around eight chops to the bar, then to use a DDL to delay a copy of
the chopped signal so that the repeats fall exactly into the gaps
created by the gating of the original part. Figure 2 should make it
clear how this is done. What you hear is a kind of chattering effect as
the repeated sections are joined up. The intelligibility is, of course,
pretty poor, as half the signal has been discarded and the other half
doubled up, but it makes for an interesting interlude. Pan the original
gated signal and its delay to opposite sides for an even more dramatic
effect.

If you're feeling adventurous, you
could set up two or more gates triggered in such a way that each gate is
only on for certain beats of a bar. Arrange your trigger material so
that only one gate is open at a time, but with one of the gates open on
each beat. This will require the generation of two or more MIDI note
control tracks or, if the gate is being triggered from a synth, you'll
need two or more different outputs to feed the gate key inputs. Finally,
feed the same signal into all three gates, but then apply a different
effect to each gate output. For example, use heavy flanging on the first
output, distortion on another and perhaps an envelope-following filter
on the third. When the outputs are heard together, you'll hear all the
differently effected beats spliced together. Figure 3 shows this
technique using three gates. If the gates click during the transitions,
lengthen the attack and release times slightly, but otherwise use the
fastest settings for the cleanest chopping.

Taking the chopping up idea even
further leads us into the murky terrain of granular synthesis and
processing, where audio is sliced into extremely short pieces that are
then joined back up in different orders to create new sounds. This
requires special software, or ingenious use of a sampler —it's not
something you can knock up using a multi-effects box, certainly, and in
many cases the process isn't even real-time. I must admit that I've yet
to hear anything musically worthwhile from processing of this kind, but
if I discover anything, you'll be the first to hear about it!

That's probably enough weirdness for
this month, but don't relax yet, as there's more to come next time when
I'll be concentrating on filtering and 'time travel'.

Monday, December 23, 2013

Sound Techniques

Technique : Effects / Processing

Modern digital effects units
always include emulations of analogue effects such as tape delay and
flanging -- but none of them ever seem quite like the real thing. Paul White explains how these vintage effects worked, and offers insight into how our modern attempts could be made more accurate.We
live in a digital age, but you can't go into a studio without hearing
stories about how good things use to sound in the old days. But were the
old electromechanical effects really that good, or have we fallen foul
of the 'nostalgia isn't what it used to be' syndrome? Certainly some old
effects were noisy and unreliable, but in most cases, there was an
element to the sound that the human ear found especially pleasing in a
musical context. Sometimes the reasons are fairly obvious, but other
times the magic element remains elusive, nowhere more so than in the
case of tape flanging or phasing.

Tape Phasing & Flanging

Tape phasing, commonly known as tape
flanging, is a unique effect, and though some digital flangers have
managed to approximate it, I've yet to hear a truly convincing
emulation. If you've never experimented with tape flanging, the effect
is created by running two identical copies of the same recording on
open-reel analogue recorders (usually in mono) and then summing the two
outputs together via a mixer at exactly the same levels. The two
recordings are started together -- a hit-and-miss business at the best
of times -- then the speed of one of the machines is slowed slightly by
using hand pressure on the tape reel. The idea is not to get so far
behind that you can hear a tangible ADT-style delay, but simply to
produce a comb filtering effect. Comb filtering occurs by virtue of the
addition and subtraction of frequencies that end up being in-phase or
out-of-phase as determined by the delay time. Whichever machine is
leading is then slowed down so that the delay decreases until the point
where the other machine takes the lead. As the relative delay between
the two tapes changes and finally passes through zero, the familiar
whooshing effect is created as the comb filter sweeps through different
frequencies in the source material. And so it continues with the leading
machine being slowed manually so that the two recordings drift in and
out of phase with each other. Because tape flanging is literally hands
on, the effect is different every time. Some of the more sophisticated
studios used electronic speed control instead of hand braking to create
the effect.

Modern flangers seek to emulate this
effect by digitally delaying one signal relative to another (by just a
few milliseconds), then modulating the delay time using a low-frequency
oscillator, or LFO. To make the effect stronger, some of the output is
fed back to the input, which adds resonance to the comb-filtering
effect. Note that with tape flanging, no feedback is used. Artificial
flanging of this kind sounds different to tape flanging for a number of
reasons -- the LFO-controlled modulation is regular, feedback is used to
add depth to the effect and the delay between the two signals never
passes through zero, as it does when two tape machines are used.

A refinement of this method is to
delay one signal by a very small amount (say 5mS), then modulate the
delay of the other signal path so that it slowly changes from less than
5mS to more than 5mS. This provides the 'through zero' element of the
effect but does nothing to break the regularity of the modulation unless
the delay time is adjusted by hand. Furthermore, if feedback is
applied, it doesn't create the desired effect, as the feedback-induced
resonance will be a function of the whole DDL delay time, whereas
the comb-filtering effect itself is related to the difference between
the two delay times. A simple setup for through-zero flanging is shown
in Figure 1 alongside the original tape-based arrangement.

In theory, it should be possible to
emulate tape flanging much more closely by using techniques such as
physical modelling. For example, one reason the effect sounds the way it
does with analogue tape machines is that analogue machines don't have
the precise phase response of a digital system. For example, put a 1kHz
square wave into a digital recorder or effects processor and what comes
out will be recognisable as a square wave. Not so with analogue tape --
the necessary frequencies are all there, but because of phase shifts in
the electronic and magnetic components of the system, their time
relationship is disturbed, which is why the waveform looks very
different to the original. The simulations might be significantly closer
if we were able to emulate this smearing before delaying the signals,
as well as introducing more randomisation into the modulation.

Plate Reverb

During the '60s and '70s, most
artificial reverb used on recordings was generated using a reverb plate,
sometimes called (inaccurately) an echo plate.The reverb plate is an
ingeniously simple device, but it takes a lot of tweaking at the design
stage to get it sounding right. Plates work by suspending a thin sheet
of metal under tension within a rigid frame via springs or clamps
attached to the corners. A transducer similar to the voice-coil of a
cone loudspeaker is used to inject audio energy into the plate and two
or more contact mics fixed to the surface of the plate then pick up the
vibrations inside it and feed them to preamps connected to the console
effect returns. By feeding the different contact mics to left and right
channels, a pseudo stereo reverb output is created.

Because metal plates have a tendency
to 'ring', getting the plate thickness, size, material and tension right
is quite an art, and some pre- and/or post-reverb EQ is invariably
needed to fine-tune the sound. Furthermore, because the plate is very
sensitive to external sounds and vibrations, it has to be mounted in a
soundproof box, ideally on shockmounts.

Unlike digital reverbs, which have
innumerable adjustable parameters, the plate reverb relies purely on EQ
for tonality and physical damping for decay-time control (usually via a
motorised felt pad). Pre-delay was often added by using an open-reel
tape machine on the input, and replaying the input signal via the replay
head to exploit the time gap between the record and replay head. You
just had to hope the tape reel didn't run out during a mix...

Because a typical plate may only be
between one and two square meters, and because sound travels much faster
in metal than in air, the reflection density within the plate builds up
very quickly following an impulse. The sound from the input transducer
spreads rapidly across the surface of the plate in all directions until
it encounters the edges of the plate, whereupon it is reflected and re-reflected
back into the plate. To get the most random reflection build-up, it's
best to have the transducer mounted a little way off-centre, and to get a
wide stereo image, the two pickups are sited at slightly different
distances from the plate edge, as shown in Figure 3 (right).

The characteristic plate sound is
bright and extremely dense, with little or no impression of individual
early reflections. The reverb builds very quickly and decays smoothly
with a maximum undamped decay time of several seconds. Digital reverbs
can provide a reasonable emulation of the coloration and envelope of a
plate reverb unit, but only the more processor-intensive models produce
the speed of reflection density build-up required to be truly
convincing. Plate reverb was very popular for vocal and drum treatments,
and though digital reverb has largely replaced it, many purists still
prefer the 'real' plate sound for certain applications.

Spring Reverb

Those who couldn't afford plate
reverbs used spring reverb -- a system still used today in guitar
combos. The physical principles are similar to those of the plate
reverb, except that sound is injected into one end of a loosely coiled
metal spring rather than a metal plate, usually via a small magnetic
transducer. A pickup transducer at the other end picks up the sound as
it reflects back and forth along the spring.

Springs invariably impart a metallic
coloration to the sound and they also tend to have a cyclic
characteristic as percussive sounds cause vibrations to bounce back and
forth along the spring in a fairly regular manner. Excessive input
levels cause the springs to 'twang', so some systems incorporated an
input limiter. To help even out the coloration and cyclic modulation,
it's possible to use two or more springs operating side by side, each
with slightly different mechanical characteristics and serviced by
individual transducers.
Using two similar sets of springs to treat left and right channels
produces a convincing stereo effect due to the non-correlated nature of
the two spring outputs.Like plate reverbs, their main
advantage is that they don't produce the gritty 'shattering'
early-reflection effects of digital reverberators. Figure 4 shows a
stereo spring reverb.

Tape Loop Echo

Before digital electronics and
charge-coupled delay lines (analogue echo), delay effects were
invariably created using tape-loop echo units such as the Echoplex,
Roland's famous RE201 Space Echo, or, at a somewhat lower cost, the
Watkins Copicat. They all worked on the same principle -- a loop of tape
passes around a series of heads starting with an erase head, followed
by a record head fed from the signal to be treated. Playback heads are
positioned after the record head to provide the echoes, and some of the
delayed signal is fed back into the record circuitry to create decaying
echoes. Delay time is varied by switching heads or varying the tape
speed, and models with multiple heads usually have switching systems for
setting up different delay patterns (see Figure 5).

Basic digital delays only approximate
tape-loop devices, even if multiple delay taps can be set to different
delay times. There are various reasons why tape-based systems sound so
distinctive; one of the main ones is the restricted frequency response
of a loop of tape that's been dragged over a set of tape heads thousands
of times. The tube circuitry of the original models also had a limited
bandwidth and introduced a significant amount of harmonic distortion.
This, combined with tape's tendency to saturate meant that when feedback
was used, successive echoes became less bright and more distorted,
creating a sense of the sound receding into the distance. On top of
that, there was instability in the tape path caused by worn rubber pinch
rollers that translated into low-level pitch modulation. As with
distortion, this type of modulation becomes cumulative when feedback is
used. Another feature that most of us would rather forget was tape
noise, but there's no denying that a good tape-echo unit has a much more
'organic' sound than even the best digital emulations.

In conclusion, it is odd that vintage
equipment is now valued because of its sonic 'imperfections', yet none
of these were designed in deliberately. Modelling is still in its
infancy, however, and I don't expect it to be long before vintage
effects can be replicated much more accurately. Perhaps then we will be
able to move on to inventing the future rather than striving to recreate
the past.

Tips & Tricks

Technique : Effects / ProcessingPart
2: Gates are far more than just problem solvers for reducing spill and
noise. They can be used to add punch to drum sounds, put rhythmic
interest into sustained parts or even as mixing automation, as Paul
White explains. Additional material by Mike Senior. This is the last article in a two-part series. Read Part 1.

Last month I explained that there are
good reasons why fully featured gates have more sockets and knobs than
you'd expect — though simple audio I/O and a threshold control might be
enough to deal with the simplest of gating tasks, there are many
situations where such facilities would prove inadequate. Now I'm going
to consider how best to apply such features to your studio tasks, and
how the host of gating controls can transform a useful studio
problem-solver into a versatile and creative mixing tool.

To Gate Or Not To Gate?

As a starting point, you have to
decide exactly when in the recording process to gate your audio signals.
A gate which is badly set up can completely ruin a signal, so if at all
possible it's best to gate when mixing rather than when recording. If
you must gate when recording, then double-check that the settings you
have chosen at least don't cause any wanted audio to be muted in the
part you're working on.

When mixing a multitrack
recording, it is common practice to employ several gates, even if
individual tracks don't seem too noisy on their own. This is because
noise is cumulative, with every playback track of a multitrack recording
contributing to the general level of noise arriving at the mix buss, so
tracks are always best muted when not in use. If you are working with a
digital system you can edit out any regions containing only noise, or
you can simply mute tracks whenever they're unused with mixer
automation, if you have it. However, the fact that gates can be set up
to perform this function automatically often means that gating proves a
more elegant solution.

If you are using any equalisation or
compression, you'll normally want to use the gate first in the signal
chain. The reason for this is that successful gating usually requires
adjustment of the gate's response to the exact levels and timbres of the
wanted and unwanted parts of your audio signal. Tweaking a pre-gate
equaliser could easily mean that you have to also re-tweak your gate's
threshold and filtering controls. Compressing a signal before gating it
can make reliable triggering even more difficult to achieve — the
compressor will make the unwanted signal a 'moving target' by modulating
its level.

Generally it gives you more
flexibility at mixdown if you record sounds without any effects such as
delay and reverb, as the levels of such effects often need to be
assessed in the context of the complete track. However, if you have
recorded any sound with such effects, you should be careful when gating
that track, so as not to shorten or modify the decay characteristics
unnaturally. Unless, that is, you want that effect...

Getting Creative With Duckers

Some gates with an
external side-chain also allow you to engage a ducker mode, where the
action of the gate is reversed — when a signal exceeds the threshold the
gate closes, rather than opens. The gating controls act in exactly the
same way as before, except that the release time refers to the speed at
which the gate opens and the attack time to the speed at which it
closes.The traditional use
for duckers is at radio stations, where they reduce the music level
while the DJ is speaking, allowing the voice to be heard more easily.
However, they also have a number of more creative uses in the studio.
For a start, you can feed the same signal to a gate and to a ducker,
panning them to opposite sides of the stereo spectrum, to create an
auto-panning effect related to the level of the sound — some gates allow
you to stereo link two gating channels even when one is switched to
ducker mode, which can make setting this up much easier.Duckers can also make
mixing easier, allowing you to make individual sounds more audible
without seriously affecting your overall balance. For example, if your
kick drum isn't cutting through enough, why not duck (using a low range
setting) some of the sounds you feel are obscuring it. Every time a kick
drum comes along, the conflicting parts will therefore be momentarily
reduced in level. You might be surprised at how effective this technique
can be with percussive sounds in particular, as you only need a very
brief ducking action to let a little more attack transient through.Similarly, you can
create extra dynamic motion in a part by ducking it several decibels at
the beginning of the bar with a MIDI click. If you then adjust the
release time so that the sound reaches its maximum level only at the end
of each bar, this sound will consistently draw you towards the next
downbeat. Though this works particularly nicely with rhythm guitar and
rhythmic synth arpeggiations, it's also worth trying with pads as well.

Clean Triggering

Traditional gating is all about the
reduction of unwanted noise or spill, and the first thing to tackle with
such problems is getting the gate to trigger exactly when you want it
to. The first thing to reach for will be the Threshold control, which
you should, in general, set as low as possible while avoiding false
triggering. It can help here if you use fast times for the gating
envelope's attack and release, as these will allow you to see exactly
how the gate is triggering — though many gates also have useful LEDs
which indicate the current action of the gate.

There are, though,
occasions where increasing the gate threshold beyond the optimal
noise-removal level can be desirable. With a fast attack time, a higher
threshold causes the gate to open abruptly only when the signal has
already reached a high level, adding a useful degree of punch to soggy
kick drum sounds. Do make sure none of the quieter hits are being
missed, though.

You will often be able to get
satisfactory triggering with no more work than this, particularly if the
unwanted signal is noise or mains hum. However, if background noise is
particularly obtrusive and localised in the frequency range, then you'll
probably need to dial in some side-chain filtering to get the gate
triggering cleanly. For example, the fundamental frequency of mains hum
and its first couple of harmonics could be removed by filtering the
side-chain below about 200Hz, whereas many electronic buzzes can be
dealt with by removing a little top end.

Side-chain filtering will almost
certainly be necessary when dealing with spill from other instruments,
particularly when gating a multitrack recording of a drummer (see my
article on drum mixing in SOS February 2001 for more on this).
However, it's worth bearing in mind that excessive high-frequency
filtering might have a fairly noticeable effect on how quickly the gate
responds to attack transients, even when dealing with basses and kick
drums.

If triggering is still proving a
problem, even when you've experimented with the gating threshold and the
side-chain filters, then it could be worth bringing hysteresis and
hold-time controls to bear on the problem as well, as they can often
help the gate respond more reliably. However, if nothing can get your
gate triggering absolutely how you want then you should try at least to
make sure that no parts of the wanted sound are lost. It's always better
to accept some noise or spill, rather than taking the risk of losing
part of your recording.

If you're using only a simple gate,
some false triggering is often unavoidable, and gating out spill on
individual drum tracks can be particularly demanding, even when you have
relatively sophisticated gates at your disposal. Usually this won't
cause you any serious problems, as any spill that occasionally gets past
the gate will normally be obscured by other instruments or ambience.
However, in some cases the sound of spill being gated on and off can be
more distracting than more continuous background noise. In cases like
these, try setting the gate's range control to simply drop the noise by a
few decibels rather than muting it altogether. Also, remember that
adding effects such as reverb and delay at mixdown will often help
disguise any small gating irregularities.

Creative Uses Of MIDI Gates

There are a certain
gates which provide ports for external triggering over MIDI, and which
can also generate MIDI events whenever they are triggered
conventionally. Such devices can make a number of creative gating
applications a lot easier, because you can program the gate action
directly into your sequencer, rather than having to send MIDI to a sound
module connected to the gate's key input.However, the ability
to trigger MIDI notes from audio events opens up a whole host of
possibilities for new drum sounds. While spill levels make it difficult
to replace a single sound within a live drum kit with an utterly
different one, that doesn't stop you from layering a new sound over
those that are already there. It effectively allows you to add any
missing element to a recorded drum sound by triggering an audio sample
alongside — often a much more elegant fix than heavy-handed EQ. If you
have one of the more advanced MIDI gating units, it will even generate
MIDI velocity information from incoming audio events, which allows the
sample's volume to track that of a drum performance more accurately.The main thing to bear
in mind when doing this type of sound replacement is that such a
sample-triggering system will not be instantaneous, especially over
MIDI. For a start, MIDI gates always take a finite time to generate a
MIDI message once the signal exceeds the threshold and, secondly,
samplers and sound modules take a few milliseconds to sound after they
have received this MIDI command. The best way around this, of course, is
to record (unquantised) the messages received from the MIDI gate into a
sequencer that's sync'ed to the original recording. From there you can
shift the data earlier, a little at a time, until you have compensated
for any delay — many of the software sequencers have a negative delay
control for each track which allows you to do this non-destructively
during playback. Recording into a sequencer also has the advantage that
you can edit out any false notes caused by unintentional triggering of
the gate.

Tweaking The Envelope

Once you're sure that the gate is
triggering reliably at the right time, then you'll need to consider how
the gating sounds. The first control you're likely to have to reach for
now is the release-time control. This will normally need to be adjusted
so that the natural decay of the sound being gated is disturbed as
little as possible. If set too short, the end of the sound will be
unnaturally truncated, whereas if the release time is too long, you'll
hear noise or spill dying away after the wanted sound has finished. The
hold-time control can sometimes be tweaked to help solve any
particularly difficult problems.

For normal applications, attack times
should be as fast as possible, particularly for percussive sounds. The
only thing to bear in mind is that extremely fast gate settings can
cause clicks when the gate opens — while this can be a boon for drums,
it can be problematic elsewhere. On the other hand, you might find that
your gate doesn't seem able to react fast enough to your drum sounds, in
which case you may have to reduce the gating range to allow it to open
more quickly.

Pushing The Envelope

Normally the gate is a problem solver,
removing elements of your audio tracks which you'd prefer not to hear.
However, gates have many uses which are far more creative than this.
Threshold and time controls in particular can reshape sounds in
interesting ways. For example, fast-attack sounds can be made to sound
almost 'bowed' if their gating envelope is given a long attack time.
Drum envelopes can be tweaked for more or less attack, and you can make
them seem almost synthetic if you gate with a very short release time.
Strummed acoustics, rhythm guitars and rhythmic bass parts can also
often be made more punchy by getting the gate to trigger on every strum
or note (rolling off a lot of low end in the side-chain can help here)
and then setting the range control for only a few decibels of gain
reduction — any and all time controls available can be used to tweak the
resulting envelope modulation.

The 'chattering' effect which can
occur when signals linger around the gating threshold can also be used
creatively as a new type distortion process. If you heavily compress
your audio signal before it reaches the gate, and then set up the gate
with its fastest time settings and with the minimum of hysteresis, you
can often achieve quite consistent periods of extremely rapid
chattering. The distortion that this causes can be extremely harsh, but
can also be softened into something much more useable with the range or
time controls. Bass sounds respond particularly well to this technique,
because the waveform often moves slowly enough that the gate can
actually modify the individual cycles of the waveform itself — great for
adding a little edge to the sound.

Triggering From Your Sequencer

As nifty as the above processes can be, the real
creative potential of gating becomes available when an external
side-chain input (or 'key' input) is provided. This is because this
input allows you to control the gating action of one signal from the
level envelope of another, a technique which I introduced last month as a
way to improve the rhythmic tightness of bass instruments and backing
vocals.

However, a more up-to-date use for
gates is for chopping up sections of audio in a rhythmic manner — an
effect used by the Prodigy on some of their guitar parts. (Garbage even
gate the entire track at several points on the opening track of their
eponymous debut album!) This is a very simple effect to achieve: the
signal being chopped is passed through the gate in the usual way but the
gate is externally triggered from a rhythmic sound fed in via the key
input. If a sustained sound (such as an organ patch) is used to trigger
the gate, the note duration can be used to control the the duration of
each segment of gated sound. However, if you are triggering with a drum
sound, the gate's hold-time control can also be used. Some hardware and
software gates can be controlled directly via MIDI, and these may be
able to achieve this effect more easily — see the 'Creative Uses Of MIDI
Gates' box.

But the usefulness of the above
triggering example doesn't end here. If you lengthen the attack and
release times and reduce the gating range, you can created a rhythmic
tremolo, rather than hard gating. What's more, if you feed the same
signal to a second gate, triggered with a delayed version of the first
gate's key input signal, you can pan these two gates to opposite sides
of the stereo image to implement auto-panning. (Note that this can also
be achieved by using a gate and a ducker, with the advantage that you
don't have to delay one of the trigger signals — see the 'Duckers' box.)
Alternatively, you could feed the two gates to a couple of different
effects processors to create interesting rhythmic modulation treatments.
And there's no need to stop there. If you have enough gates and
triggering sources, there need be no limit to the effects wierdness you
can generate — check out my article on extreme effects in SOS November 2000 if you want to go further with this.

Downward Expansion And The True Upward Expander

There are two types of expander, but the most
common type is that which functions similarly to a gate, reducing the
volume of signals which are below a threshold level. The way an expander
of this type differs from a gate, however, is in the way it
reduces the gain of sub-threshold signals: where a gate simply applies a
fixed amount of reduction (set by the range control), the expander
reduces gain proportionally (according to the setting of a ratio
control). For example, a 1:3
expander ratio would mean that for every decibel the input fell below
the threshold, the output would fall by 3dB. Naturally, once the signal
exceeds the threshold, the gain is returned to normal. Though signals
expanded in this way can sound quite odd, the resultant level changes
can still seem less abrupt than those produced by simple gating,
especially when fast attack and release times are required.The other type of
expander, often called a 'true upward expander', is less common. Rather
than expanding the dynamic range below the threshold, it expands above
the threshold, once again according to a ratio set by the user. The
sound of this type of expansion is subtly different to that of the more
common type, and can be particularly useful in making attack transients
more prominent. If the threshold is set relatively high, it can also
reverse some of the detrimental effects of heavy-handed mix compression,
allowing you to rescue otherwise unusable sections of audio at the
mastering stage. However, this process is by no means infallible, so
it's still better to err on the cautious side when processing overall
dynamics during mixdown.

Mix & Match

While using sequenced MIDI sound
sources to trigger gates can be fun, you don't have to use artificially
generated trigger signals to get creative with gating. One of the joys
of mixing in the analogue domain is that every track in a mix can be
used as a trigger source for any number of gates operating on other
tracks — a flexibility which is still surprisingly rare in digital
systems!

One of the most famous (some might say
infamous) effects that can be produced in this way is the '80s drum
sound with gated reverb, as evidenced on the Phil Collins classic 'In
The Air Tonight'. If an extremely ambient drum sound is gated with a
high threshold, fast attack and release times, and a longer hold time,
each drum hit becomes a concentrated burst of sound and can therefore
seem more powerful. Figure 2 shows how to set this up. If you are
wanting to create this effect with natural ambience, the effect would
work best in a large, live room. A concrete stairwell will produce good
results as long as the mics don't pick up the neighbours complaining!
However, the gated reverb effect can also easily be achieved using an
artificial reverb unit — simply trigger a gate on the reverb return with
signals sent from the aux sends of individual drum channels. For more
characteristic results, compress the reverb return.

Though the gated reverb effect on
drums was done to death in the '80s, that doesn't stop you using it to
thicken other sounds — both rock guitars and backing vocals can respond
well to this. In fact, the lead vocal for David Bowie's 'Heroes' was
apparently recorded with three different mics — one close, one a few
feet away and the third at the other end of the room — the latter two of
which were gated with different thresholds in order to introduce
increasing amounts of room ambience as Bowie sang louder. And remember
that delaying the gate's side-chain trigger signal by a beat or two can
often provide extra rhythmic interest, when gating ambience signals.

Gating one signal from another can
also help with balancing your music while mixing. For example, it can
often really help vocal intelligibility in rhythmic music, even when the
overall vocal level is quite low. If you gate the lead vocal, with the
range control set close to minimum, and trigger the gating action from a
drums submix, this will effectively mean that the vocal level rises
momentarily with each loud drum beat, making it less likely that it will
be masked out. It even helps to counteract the possibly detrimental
effect on the vocal level of later mix compression. This trick can be
applied in a host of other situations as well, wherever different sound
sources are fighting for space in the mix.

Shut The Gate Behind You...

Gates are one of the most useful of
the studio workhorses, particularly when recording live drums and rock
bands. However, it is a shame to use them simply to reduce noise and
spill when they are capable of so much more than this. Modifying the
amplitude envelope of a sound subtly can ease the mixing process, while
drastic measures can be taken for a number of special effects. And, once
you start experimenting with external side-chain signals, the creative
possibilities further multiply.